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Renaming bandwidth to bitrate in webrtcvoiceengine.

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"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.

This is to remove the confusion inside webrtcvoiceengine

BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org 2014-10-29 02:27:08 +00:00
parent ffeaeed8c1
commit 2623695dfb
2 changed files with 27 additions and 26 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

47
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -495,7 +495,7 @@ void WebRtcVoiceEngine::ConstructCodecs() {
ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
LOG(LS_INFO) << ToString(codec);
if (IsIsac(codec)) {
// Indicate auto-bandwidth in signaling.
// Indicate auto-bitrate in signaling.
codec.bitrate = 0;
}
if (IsOpus(codec)) {
@ -1227,7 +1227,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
// Apply codec-specific settings.
if (IsIsac(codec)) {
// If ISAC and an explicit bitrate is not specified,
// enable auto bandwidth adjustment.
// enable auto bitrate adjustment.
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
@ -1792,8 +1792,8 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
engine->CreateMediaVoiceChannel()),
send_bw_setting_(false),
send_bw_bps_(0),
send_bitrate_setting_(false),
send_bitrate_bps_(0),
options_(),
dtmf_allowed_(false),
desired_playout_(false),
@ -2028,9 +2028,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
bool nack_enabled = nack_enabled_;
bool enable_codec_fec = false;
// max_playback_rate <= 0 will not trigger setting of maximum encoding
// bandwidth.
int max_playback_rate = 0;
int opus_max_playback_rate = 0;
// Set send codec (the first non-telephone-event/CN codec)
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
@ -2048,7 +2046,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
continue;
}
// We'll use the first codec in the list to actually send audio data.
// Be sure to use the payload type requested by the remote side.
// "red", for RED audio, is a special case where the actual codec to be
@ -2080,7 +2077,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// For Opus as the send codec, we are to enable inband FEC if requested
// and set maximum playback rate.
if (IsOpus(*it)) {
GetOpusConfig(*it, &send_codec, &enable_codec_fec, &max_playback_rate);
GetOpusConfig(*it, &send_codec, &enable_codec_fec,
&opus_max_playback_rate);
}
}
found_send_codec = true;
@ -2116,15 +2114,16 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
}
// maxplaybackrate should be set after SetSendCodec.
if (max_playback_rate > 0) {
// If opus_max_playback_rate <= 0, the default maximum playback rate of 48 kHz
// will be used.
if (opus_max_playback_rate > 0) {
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
<< max_playback_rate
<< opus_max_playback_rate
<< " Hz on channel "
<< channel;
#ifdef USE_WEBRTC_DEV_BRANCH
// (max_playback_rate + 1) >> 1 is to obtain ceil(max_playback_rate / 2.0).
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
channel, max_playback_rate) == -1) {
channel, opus_max_playback_rate) == -1) {
LOG(LS_WARNING) << "Could not set maximum playback rate.";
}
#endif
@ -2133,8 +2132,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
if (send_bw_setting_) {
SetSendBandwidthInternal(send_bw_bps_);
if (send_bitrate_setting_) {
SetSendBitrateInternal(send_bitrate_bps_);
}
// Loop through the codecs list again to config the telephone-event/CN codec.
@ -3187,25 +3186,27 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
return true;
}
// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
// SetMaxSendBitrate() in future.
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth.";
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
return SetSendBandwidthInternal(bps);
return SetSendBitrateInternal(bps);
}
bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBandwidthInternal.";
bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
send_bw_setting_ = true;
send_bw_bps_ = bps;
send_bitrate_setting_ = true;
send_bitrate_bps_ = bps;
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet. "
<< "The send bandwidth setting will be applied later.";
<< "The send bitrate setting will be applied later.";
return true;
}
// Bandwidth is auto by default.
// Bitrate is auto by default.
// TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
// SetMaxSendBandwith(0), the second call removes the previous limit.
if (bps <= 0)

6
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -435,7 +435,7 @@ class WebRtcVoiceMediaChannel
return channel_id == voe_channel();
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
bool SetSendBandwidthInternal(int bps);
bool SetSendBitrateInternal(int bps);
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
@ -453,8 +453,8 @@ class WebRtcVoiceMediaChannel
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
bool send_bw_setting_;
int send_bw_bps_;
bool send_bitrate_setting_;
int send_bitrate_bps_;
AudioOptions options_;
bool dtmf_allowed_;
bool desired_playout_;